Saturday, 23 March 2013

Imagine, audiophiles using an "archaic" (circa 1994) game console as a CD player for thousands of dollars worth of expensive amps and speakers downstream!

That Stereophile article reviewed the first PS1 version which was the SCPH-1001. Instead of that older model, what I have here is the slightly later SCPH-5501; said to be superior for audio by some PS1 aficionados!

Here's an interesting post from a fellow on the Steve Hoffman message board going by the alias "rhing":

I recently purchased a used Playstation1 Model SCPH-5501 from a
local video game store for about $25. I bought it based on the fact that
it uses the same Asahi Kasei AK4309AVM DAC as Model SCPH-1001. The most
obvious difference between Models SCPH-1001 and SCPH-5501 is that Model
SCPH-1001 is the only Playstation 1 with RCA stereo audio output jacks.
For some people, this is important to getting the ultimate performance
from a Playstation 1. Model SCPH-5501 outputs stereo audio through a
12-pin Sony A/V Multiport jack that uses an A/V breakout cable with RCA
jacks for Right Audio (red), Left Audio (white) and Composite Video
(yellow). In addition to the RCA jacks on the rear, Model SCPH-1001 also
has the same Sony A/V Multiport jack. The benefit to using the A/V
Multiport jack is that there are no cheap NJM2100 op amps in the signal
path between the DAC and the output. The RCA jacks are connected through
a pair of op amp buffers that can adversely effect the sound quality.
In fact, most Playstation 1 audiophile modders bypass the op amp
buffers, or use the A/V Multiport to get the same effect of better
sound. Other improvements were incorporated in the later Model
SCPH-5501:1) An improved Nichicon SMPS that generated less heat that could distort
adjoining plastic components such as the lid and chassis that could
lead to mistracking or binding of a spinning CD disk.2) Positioned the laser assembly away from the power supply to reduce heat damage and RFI noise.3) Implemented an auto-biasing feature for proper tracking. Model
SCPH-1001 requires manual biasing of the laser circuitry to maintain
tracking.

...

He goes on comparing the PS1 with other gear he has. Good read. I can't verify the comments but it certainly sounds well thought out and worth considering. For those wondering, the AK4309 is a 1-bit delta sigma "bit stream" DAC.

So, without further ado, here it is:

Notice the lack of RCA direct outputs at the back; there's a supplied multi-AV cable with RCA out instead.

As you can see, for convenience in comparison, I included the 16/44 results for a number of the other DAC/streamer units I measured over the last while.

Compared to the rest, the PS1 is clearly outmatched in terms of dynamic range. These measurements indicate that it's capable of around 15-bit dynamic range. At first, I thought it may be the fact that this is a CD player with all kinds of electronics in there perhaps lowering the dynamic range. However, when you have a look at the AK4309 datasheet, indeed the rated dynamic range for this part is 'only' 90dB.

Frequency response:

Some slight deviance from flat response curve up above 3kHz - unlikely to be noticeable through speakers and room interactions. Note the green plot for the Muse Mini TDA1543x4 for comparison to show what a typical NOS DAC measures like. Old school early 90's 1-bit delta sigma vs. NOS! :-)

THD:

Since the Transporter uses the newer generation AKM DAC (AK4396), here are the THD plots in comparison. Obviously, the Transporter is capable of measurably lower noise level with a cleaner graph notably above 10kHz.

In comparison, the TDA1543 (NOS DAC) measures much worse with numerous harmonics:

There you go, the Sony Playstation 1 SCPH-5501 measured as a CD player. I can't say how it sounds compared to the even older "first-version" SCPH-1001 but as I quoted above, there are reasons to believe this version should perform better. Last night, for some late-night R&R, I listened with this unit through my AUNE X1 as headphone amp with Sennheiser HD800 headphones. Tunes on tap: Muddy Waters "My Home Is In The Delta" (downsampled Classic Records HDAD), Cat Stevens "Wild World" (2011 Analogue Productions), Nat King Cole "The Very Thought Of You" (2010 Analogue Productions), Akon "I Wanna Love You", eRa "The Mass", Joe Satriani "Crowd Chant", Jheena Lodwick "It's Now Or Never", Stephen Layton & Britten Sinfonia "For Unto Us A Child Is Born" from Handel's Messiah. They all sound great through the headphones. Nice details throughout, bass nice and deep on "I Wanna Love You", no accentuated sibilance with female vocals (that Jheena Lodwick track can be nasty), "The Mass" may have sounded a bit congested during the louder & more complex segments but really nothing to detract from enjoyment. I do not believe the measured 'limited' dynamic range negatively affected enjoyment at normal listening levels at all. Remember that ~90dB dynamic range is better than the majority of analogue sources.

Clearly compared to modern DAC's, the PS1 has inferior noise performance and concomitantly lower dynamic range. The other measurements like THD and IMD are respectable and jitter is not of concern. If we look at the TDA1543 DAC unit (DAC chip designed around the same era), the NOS unit has better dynamic range but with much more THD and IMD along with the typical high frequency roll-off at 44kHz sampling rate (of course you could feed the NOS upsampled 88/96kHz data to smooth this out). Between the two ultimately, it's one of subjective preference, especially for a "colored" DAC like the TDA1543 NOS (I've yet to measure a tube analogue output DAC). Personally, I'm of the "technically perfect" camp and would probably pick the PS1 over the TDA1543 even if a bit noisier - reasonably low level of noise like this is generally less objectionable than distortion.

I don't have any digital gear from the late 80's left, but by the mid-90's, "CD players" like this one sound and measure fine (though far from great in the case of the PS1 here)...

Monday, 11 March 2013

Hey guys, remember the SACD vs. DVD-A "war" back in the early-mid 2000's?

In the heat of the battle, it was nice that a few manufacturers gave us these "universal" DVD players that could handle both competing formats. Pioneer was one of them and in mid-2003, released to the world the very reasonably priced DV-563A (~$200). A year or so later, this model under consideration, the DV-588A was released. I remember browsing around Future Shop (owned by BestBuy these days) in 2005 to pick up this unit since my previous Panasonic DVD player failed on me. At <$200, I figured I couldn't go wrong since this would 'in a pinch' also play my small collection of DVD-A and larger collection of SACD's.

The innards of this player revolves around the MediaTek MT1389EE SoC chipset which graced many <$400 players back in the day (including the Oppo DV-981HD and Sony NS955 [chip apparently relabeled as Sony CXD9804R]). Although it has been said that this chip is capable of "PureDSD", I do not believe any of the lower priced units operated in this mode because multichannel and bass management were performed in the PCM domain. As a result, the 1-bit DSD (actually DSD64 for the 64 x 44.1kHz = 2.8224MHz SACD sampling rate) in these machines are converted to 24/88 prior to conversion by the DAC (supposedly a BB DAC is used in this unit).

Okay, so far not too bad for a budget player... It can benefit from 24-bit audio with a dynamic range around 17-bits. Here's the 24/192 frequency response:

And here's the noise level (useful when we look at the SACD graph) - reasonably flat to about 50kHz:

As usual, I plotted the Dunn Jitter Test (24/48):

Quite jittery as you can see with a number of sidebands congregated +/- 1kHz around the 12kHz main signal.

Now as I mentioned above, this player is capable of SACD-R playback. So, I downloaded KORG AudioGate and went about converting the 24/192 PCM RightMark test and calibration signals and 24/48 Dunn J-Test into DSD64. With the help of a friend who's into this stuff, we got the tracks authored onto a SACD-R for testing.

Here's what RightMark looks like playing the SACD-R:

Not good... Just marginally better than DVD-A 16/44.

Lets have a look at the frequency response curve:

Not unexpectedly, the fact that it's resampled to 24/88 leads to an earlier cut off in frequency response than the DVD-A 24/192 spectrum above. I wonder if AudioGate is imposing a low pass filter starting around 30kHz to create that early steep drop off...

Noise spectrum:

Note that extra "bump" at 25 to 44kHz thanks to the DSD noise shaping (absent in the PCM noise graph above) - remember that this is a log scale so it's all scrunched up in the corner. The fact that the noise gets truncated at ~44kHz makes sense for the 24/88 PCM conversion, that's why I wondered above if AudioGate is cutting off the frequencies of the test signal earlier at 30kHz.

Although the Dunn J-Test is invalid in terms of stimulating worse-case jitter in the DSD domain, for the heck of it here's the spectrum (again 24/48 test):

It's cleaner than the PCM graph above. Possibly the resampling from DSD to PCM done internally - like ASRC's - is cleaner than the direct PCM from the DVD transport (I don't think it has to do with corruption of the LSB jitter modulation tone since even a straight 12kHz sine wave looks very jittery in DVD-A).

Summary:
Hey, I got to measure a DVD-A / SACD player with my custom SACD-R & DVD-A disks :-)! I've often wondered how this old Pioneer compared in terms of SACD vs. DVD-A playback. Subjectively, I've always thought it sounded good but not exceptional. I never bought the same title on both DVD-A and SACD to actually compare how the two would sound. Even if I did, there would be no assurance that the mastering would be identical anyways. Although I have about 30 SACD's still, they're barely listened to compared to the computer music server and my Squeezebox players. The fact that I do have a dedicated SACD changer still (the venerable Sony SCD-CD775) also means I don't bother with the Pioneer for music playback the few times I spin a disk not for the purpose of audio ripping.

Overall, the objective data points to a 'fair' DVD-A player for its price and a 'mediocre' SACD player with dynamic range just a hair above that of an ideal 16-bit CD. Of course I'm converting my test tracks from PCM to DSD and then the player is re-converting back to PCM again so this will affect the sound quality negatively... Furthermore, I haven't played with AudioGate enough to get a sense of how good it is as a PCM to DSD converter compared to something like Weiss Saracon which is much too expensive for the casual hobbyist. Nonetheless, since RightMark is measuring dynamic range at 1kHz (should be quite good for DSD64), it should still measure better than what I got assuming the digital data conversion was done competently and the 24/88 internal conversion algorithm is good.

The question of course is how well does a good "pure DSD" SACD player measure compared to something like this budget Pioneer? Can anyone suggest a good SACD unit to try which can play back SACD-R's?

[It would have been nice to convert 24/176 test audio to DSD rather than 24/192 due to the even multiple 44kHz sampling rate. Unfortunately the EMU-0404USB seems unable to handle this sampling rate well for me. Ideally of course, DXD 24/352 would be even better!]

Thursday, 7 March 2013

I can't help it, I guess... I remain fascinated by this whole jitter issue, particularly with the belief that computer load can somehow alter the jitter/timing characteristic of a digital interface. After years of talk in the press about jitter, there's almost a mystique about this phenomenon. After all, jitter seems to be the central tenet to the concept that "bits are not just bits" because there is a timing component which somehow gets altered by the transport of said bits into the DAC for playback... As I have alluded in previous posts, software-side techniques to reduce timing errors have been reported to include:
- strip down OS'es - prevent unnecessary processes from running
- prefer simpler/older OS'es like Windows XP, or alternate OS like Linux with RT kernels
- use large memory buffers
- specialized music player software which presumably go beyond bit perfect
- play only WAV / AIFF files (ie. FLAC decoding can add jitter)

While I would not be able to test out all these assertions, I think we can logically deduce that IF timing is an issue and can be altered by computer load (whether due to OS, player software, FLAC decode, etc...), we should see some kind of anomaly with the Dunn J-Test when the computer gets busy in realtime.

(Notice a few hiccups with the realtime spectrum analyzer... Quite alot of numbers to be crunched to plot out the 131,072 point FFT while recording video and audio.)

Realize just how "basic" this setup is... I'm using just the built-in motherboard TosLink straight to the $200 DAC.

As I have shown so far in the other tests, jitter is measurable between
different interfaces. Also, electrical noise is easily measurable (for
example, see the "NOISY i7" condition with the Essence One testing). However, I am still unable to show that multitasking or running the CPU at high load has any ability to change the timing/jitter characteristics of the Dunn J-Test significantly much less to an audible degree. As far as I can tell, the jitter phenomenon is a property of the digital hardware interface itself (ie. TosLink and adaptive USB tend to be worse than coaxial, AES/EBU, and asynchronous USB). Therefore, my suspicion/belief is that so long as the computer software player is functioning properly (ie. no buffer under runs, feeding a bitperfect driver like ASIO), then there should not be any jitter issues other than the limitation of the computer-to-DAC interface (at least in this case with a modern CPU & motherboard chipset).

In an even more extreme situation, with the AUNE X1 adaptive USB interface running off a USB hub with a hard drive attached transferring 25MB/sec data plus the CPU running 100% while playing the jitter test, I have not seen deterioration in the J-Test spectrum. (I won't bore you with the J-Test graphs since they look exactly the same whether computer is idle or busy and transferring files over the USB hub!)

Realize that this finding is very good! It means that we're free to do stuff like realtime transcoding and use of fancy upsampling algorithms without fear that somehow it will deteriorate the sound and that jitter is an independent variable not affected by the audio processing itself.

If anyone has reason to believe otherwise, I'd love to have a look at the test set-up and evidence of software-induced jitter (especially if it's audible!).

--------------------Addenda:

Since I'm obsessive-compulsive and pedantic, here are a few measurements related to the above...

1. Just to show what a modern motherboard's analogue output looks like (ASrock Z77 Extreme4 motherboard "HD" sound, RealTek ALC898 codec, all "enhancements" off). Here are the measurements using a shielded phono-to-RCA cable to the E-MU 0404USB:

24/96 Frequency Response:

24/96 THD graph:

Summary - OK for 16-bit audio in terms of noise floor and dynamic range. Incapable of going beyond 16-bit dynamic range at best with 24-bit audio data. I suspect this is quite typical of on-board sound output. Notice the deterioration with 24/192.

2. Does the jitter pattern from the motherboard's analogue output itself get worse with increased CPU load?

Setup: Analogue output from ASrock Z77 Extreme4 --> shielded Phono to RCA --> E-MU 0404USB
Used the 24/48 J-Test like in the video above (that one was of course with the computer TosLink).

Conclusion: The motherboard's internal DAC is relatively jittery compared to the USB DAC's tested below. But symmetrical jitter sidebands are no different whether CPU or GPU load high. Bottom
line... I can't even seem to stimulate more jitter with the
motherboard's own internal DAC by increasing CPU or GPU load. One
consistent finding though is that bit of noise between 8-9kHz - usually
whenever I strain the GPU.

3. I alluded to the AUNE X1 adaptive USB interface put under stress. Remember that in this case I'm using a separate external USB hub (not even the direct motherboard USB port); here are the boring graphs - 16-bit J-Test since the adaptive interface is incapable of 24-bit:

Computer idle: (note the 15kHz distortion at -90dB - discovered this to be a driver issue with ASIO4All - see April 1 update... Might never have noticed this if not for testing.)

No difference... Very good graphs with minimal jitter (the spikes are primarily the 16-bit jitter modulation signal from the J-Test). Clearly the asynchronous interface is better even going through the TosLink.

This whole 'jitter' thing is getting tired and boring :-). Yawn... I'm not using any exotic or expensive gear at all and yet I can't even get jitter anomalies to show up despite the strain I've put on the CPU/GPU and USB interface.

Wednesday, 6 March 2013

A little while ago, I demonstrated that the TosLink loopback with the Behringer DEQ2496 in line worsened the Transporter's jitter measurements significantly here. Although I do not believe the extra 2ns or so of jitter was audible, I wondered if using an alternate interface than TosLink would have improved the situation. Although the Behringer doesn't have a coaxial SPDIF, it does feature the AES/EBU interface which is a digital balanced cable for me to try.

So, I got a couple of 5' Mogami W3080 + Neutrik connector cables for total ~$65 shipped to take the AES/EBU interface for a spin.

For the TosLink condition, I'm using 2x3' generic plastic TosLink cables instead of the AES/EBU.

Firstly, I wanted to make sure the analogue output remains good/unchanged:

The upper table contains 24/88 measurements - these days, it seems more hi-res is available in 24/88 (often SACD/DSD conversions), I figured it would be good to have a look to make sure it all measured well. The lower chart are the same conditions at 24/96. Note that these are the RCA output measurements so a little lower than with the XLR results posted before. Note that there is some inter-test variability compared to my previous results with the Transporter but generally we're talking <1dB difference.

As you can see, from the analogue perspective, there isn't any difference whether I'm measuring the Transporter DAC directly, or if it's running through a total of 6' TosLink or 10' AES/EBU through the Behringer DEQ2496.

Let us turn our attention to those pesky jitters. As usual, using the Dunn J-Test:

Transporter direct (16/44):

Transporter-DEQ2496 AES/EBU loopback (16/44):

Transporter-DEQ2496 TosLink loopback (16/44):

At 16/44, the Transporter is very clean for both direct and AES/EBU loopback. Most of these spikes are just J-Test modulation products from the 16-bit signal being recorded in 24-bits. As you can see, the TosLink loopback is considerably worse with more sidebands congregated around the primary 11kHz signal. You can also see that using the TosLink interface with the Transporter slightly raises the absolute noise floor in general.

The AES/EBU loopback does add a small amount of jitter to the graph but it really is quite insignificant!

In the 24-bit domain, here are the same conditions with the 24/48 J-Test:

Transporter direct (24/48):

Transporter-DEQ2496 AES/EBU loopback (24/48):

Transporter-DEQ2496 TosLink loopback (24/48):

These graphs look a bit different from the previous Transporter jitter measurements because I'm measuring RCA output (rather than the XLR from before which has a lower noise floor). Nonetheless, the results are the same in terms of jitter - TosLink is significantly worse.

Conclusion:

Well, I think that's it for my suite of Transporter measurements... The practical side realizes that jitter at these levels even with TosLink doesn't make an iota of difference playing real music. However, the audiophile seeks for "perfection" in as much as it's possible. Using AES/EBU digital cables instead of TosLink in this loopback configuration with the Behringer DEQ2496 for room-EQ does not add any significant extra jitter within the resolution limits of my test equipment. Cool.